1. Field of the Invention
The present invention relates to a method and an apparatus for characterizing an audio transmitting system and also to a method and an apparatus for setting the characteristics of an audio filter. More particularly, the invention relates to a method and an apparatus for characterizing an audio transmitting system and also to a method and an apparatus for setting the characteristics of an audio filter, both of which are required for an equalization system that simulates an ideal acoustic space in a vehicle cabin.
2. Description of the Related Art
Generally, the propagation characteristics of sound which is emitted from a speaker installed inside a vehicle and heard are very complicated. This is one of the major factors that deteriorate the reproduced sound of an audio system. In order to overcome the above drawback, an audio system which controls a sound field by using a digital filter has been proposed.
An audio system shown in FIG. 7 is formed of an audio source 100 having a radio tuner and a CD player, a control filter 102 for controlling the frequency characteristics of audio signals output from the audio source 100, and a speaker 104 for radiating audio output from the control filter 102 into a vehicle cabin. Meanwhile, a target-response setting unit 106, in which target response characteristic (impulse response) H is set, receives an audio signal from the audio source 100 and outputs the corresponding target response signal. As used hereinafter, in the specification, the uppercase C, H, W, X, and U each indicate a vector quantity, and the lowercase w, x, and u each represent a scalar quantity. A computation unit 108 calculates an error (difference) between a sound signal output from a microphone 110 installed at an audio-detecting position (listening position) of a vehicle acoustic space and the target response signal output from the target-response setting unit 106. Thus, a determination of the characteristics of the control filter 102 so as to minimize the error calculated by the computation unit 108 enables the simulation of an ideal acoustic space.
The characteristics of the control filter 102 incorporated in the above-described audio system are determined by the following two steps.
In the system illustrated in FIG. 8, the characteristics of a finite impulse response (FIR) adaptive filter 112 are first determined so as to be comparable to a vehicle cabin acoustic system C, thereby characterizing the audio transmitting system. Generally, as the adaptive algorithm for determining the characteristics of the FIR adaptive filter 112, a least mean squares (LMS) algorithm is used. According to the LMS algorithm, a vector W of each tap coefficient of the adaptive filter 112 is updated each sampling time based on the following equation:
W(n+1)=W(n)+xcexc1X(n)xcex5(n)xe2x80x83xe2x80x83(1)
where W(n) indicates a vector of a tap coefficient with respect to a time n; and X(n) represents a vector of a reference signal input into the system shown in FIG. 8. The factors W(n) and X(n) are further expressed by the following equations:
W(n)=[W(n,0),w(n,1), . . . ,w(n,Lxe2x88x921)]
X(n)=[x(n),x(nxe2x88x921), . . . ,x(nxe2x88x92L+1)]
where L indicates the number of taps of the adaptive filter 112. Moreover, in equation (1), xcexc1 indicates a step size parameter, and xcex5(n) designates an error signal. The error signal represents the difference between a detection signal dxe2x80x2 output from the microphone and an output of the adaptive filter 112. The above-described updating operation is repeated until the factor xcex5(n) becomes equal to or smaller than a predetermined value.
Subsequently, the system illustrated in FIG. 9 is constructed by using the characteristics of the audio transmitting system determined by the foregoing step 1. The characteristics of a FIR adaptive filter 114 are then determined. Design of the control filter shown in FIG. 7 is thus performed. Referring to FIG. 9, a filter 116 is formed by fixing each tap coefficient of the adaptive filter 112 determined by the foregoing step 1 and is provided with characteristics corresponding to the vehicle cabin acoustics. The target-response setting unit 106 illustrated in FIG. 9 is the same as the one used in the audio system shown in FIG. 7, and ideal target response characteristics are set in the target-response setting unit 106.
A filtered-x LMS algorithm is generally employed, as the adaptive algorithm that determines the characteristics of the adaptive filter 114. According to the filtered-x LMS algorithm, a vector W of each tap coefficient of the adaptive filter 114 is updated each sampling time based on the following equation:
W(n+1)=W(n)+xcexc2U(n)e(n)xe2x80x83xe2x80x83(2)
where W(n) represents a vector of a tap coefficient with respect to a time n; xcexc2 indicates a step size parameter; and U(n) designates a vector of a reference signal output from the filter 116 with respect to a time n. U(n) is further expressed by the following equation:
U(n)=[u(n),u(nxe2x88x921), . . . ,u(nxe2x88x92L+1)]
where L indicates the number of taps of the adaptive filter 114. Moreover, e(n) indicates an error signal representing the difference between a detection signal dxe2x80x2(n) output from the microphone and a target response signal d(n) output from the target-response setting unit 106. The above-described updating operation is repeated until the factor e(n) is equal to or smaller than a predetermined value.
The control filter 102 shown in FIG. 7 is designed by fixing the individual tap coefficients of the adaptive filter 114 which are determined by the foregoing two steps.
In this manner, the systems illustrated in FIGS. 8 and 9 including the adaptive filters 112 and 114, respectively, are constructed, and the individual tap coefficients are set by using predetermined algorithms, thereby enabling the design of the control filter 102. It is necessary, however, that the adaptive filters 112 and 114 be constantly and stably operated in order to design the control filter 102 as described above.
For stably operating the adaptive filter 112 in the system illustrated in FIG. 8, the step size parameter xcexc1 contained in equation (1) should satisfy the condition indicated by the following expression:
0 less than xcexc1, less than (2/(LE[x(n)2]))xe2x80x83xe2x80x83(3)
wherein L indicates the number of taps of the adaptive filter 112; E[ ] represents an expectation operator; and x(n) indicates a momentary value of the reference signal with respect to a time n.
Similarly, for stably operating the adaptive filter 114 in the system illustrated in FIG. 9, the step size parameter xcexc2 included in equation (2) should satisfy the condition represented by the following expression:
0 less than xcexc2 less than (2/LE[u(n)2])xe2x80x83xe2x80x83(4)
where L indicates the number of taps of the adaptive filter 114; and u(n) indicates a momentary value of the reference signal with respect to a time n.
In a typical audio system, such as the one shown in FIG. 7, it is vital that the control filter 102 be stably operated. Accordingly, the adaptive filters 112 and 114 used for designing the control filter 102 should be operated stably. Thus, the step size parameters xcexc1 and xcexc2 are required to satisfy the conditions represented by expressions (3) and (4). The factors E[x(n)2] and E[u(n)2] calculated by using expectation operators are contained in expressions (3) and (4), respectively. Thus, in order to reliably satisfy the above conditions (expressions (3) and (4)), the step size parameters xcexc1 and xcexc2 should be set by sequentially measuring the power of the reference signals x(n) and u(n). However, the addition of a structure required for measuring the power of the reference signals x(n) and u(n) increases the complexity of the systems illustrated in FIGS. 8 and 9, further causing an increase in the cost of the product.
Accordingly, in view of the above, it is an object of the present invention to provide a method for characterizing an audio transmitting system and a method for setting the characteristics of an audio filter, both of which ensure the stable operation of adaptive filters without increasing the complexity of the overall system.
In order to achieve this object, according to one aspect of the present 1invention, there is provided a method for characterizing an audio transmitting system. In this system, tap coefficients of an adaptive filter are set by using an LMS algorithm, in which case, a white noise signal is used as an input signal into the adaptive filter.
In this LMS algorithm, it is generally known that the step size parameter xcexc1 should satisfy the condition represented by expression 0 less than xcexc1 less than (2/(LE[x(n)2])) in order to stably operate the adaptive filter, where L indicates the number of taps of the adaptive filter, E[ ] represents an expectation operator, and x(n) is an input signal into the adaptive filter at a time n.
In the present invention, a white noise signal having an average power of one is used as an input signal into the adaptive filter. Accordingly, the above expression can be modified into 0 less than xcexc1 less than (2/L), and the upper limit is fixed. This eliminates the need for setting the step size xcexc1 upon sequentially calculating the average power of the input signals into the adaptive filter. Moreover, a structure which would otherwise be required can be omitted. Additionally, by setting the step size parameter xcexc1 in a fixed range, the adaptive filter can be stably operated.
According to another aspect of the present invention, there is provided an apparatus for characterizing an audio transmitting system. The apparatus includes a signal generating unit for generating a predetermined signal. A speaker radiates the predetermined signal generated by the signal generating unit into an acoustic space. A microphone is placed at a predetermined position in the acoustic space and collects sound radiated from the speaker. An adaptive filter receives the predetermined signal output from the signal generating unit. A computation unit calculates the difference between an output signal of the microphone and an output signal of the adaptive filter and outputs the difference as an error signal. A first algorithm-processing unit receives the predetermined signal output from the signal generating unit and the error signal output from the computation unit and sets a tap coefficient of the adaptive filter by using a predetermined algorithm. Thus, the apparatus determines characteristics of the audio transmitting system according to characteristics of the adaptive filter.
According to still another aspect of the present invention, there is provided a vehicle-loaded audio system incorporating the foregoing apparatus for characterizing an audio transmitting system.
According to a further aspect of the present invention, there is provided a method for setting characteristics of an audio filter. In this method, tap coefficients of an adaptive filter are set by using the filtered-x LMS algorithm, in which case, a white noise signal is used as an input signal into the adaptive filter.
In this filtered-x LMS algorithm, it is generally known that the step size parameter xcexc2 should satisfy the condition represented by expression 0 less than xcexc1 less than (2/(LE[u(n)2])) in order to stably operate the adaptive filter, where L indicates the number of taps of the adaptive filter, E[ ] represents an expectation operator, and u(n) represents a reference signal used in the filtered-x LMS algorithm at a time n.
In the present invention, a white noise signal having an average power of one is used as an input signal into the adaptive filter. Accordingly, the above expression can be modified into 0 less than xcexc2 less than (2/(LCt C)) (C represents a characteristic vector indicating an impulse response of an acoustic space), and the upper limit is fixed. This eliminates the need for setting the step size xcexc2 upon sequentially calculating the average power of the reference signals used in the filtered-x LMS algorithm. Moreover, a structure which would otherwise be required can be omitted. Additionally, by setting the step size parameter xcexc2 in a fixed range, the adaptive filter can be stably operated.
According to a further aspect of the present invention, there is provided an apparatus for setting characteristics of an audio filter. The apparatus includes a signal generating unit for generating a predetermined signal. An adaptive filter receives the predetermined signal generated by the signal generating unit. A speaker radiates an output signal of the adaptive filter into an acoustic space. A microphone is placed at a predetermined position in the acoustic space and collects sound radiated from the speaker. A target-response setting unit, in which predetermined target response characteristics are set, outputs a target response signal corresponding to the predetermined signal input into the target-response setting unit. A computation unit calculates the difference between an output signal of the microphone and the target response signal output from the target-response setting unit and outputs the difference as an error signal. A second algorithm-processing unit receives the predetermined signal output from the signal generating unit and the error signal output from the computation unit and sets a tap coefficient of the adaptive filter by using a predetermined algorithm. Thus, the apparatus sets characteristics of the audio filter according to characteristics of the adaptive filter.
According to a yet further aspect of the present invention, there is provided a vehicle-loaded audio system incorporating the foregoing apparatus for setting the characteristics of an audio filter.